On Thu, 2003-09-18 at 08:36, denzel-infotechs wrote: > hi! > Infact the problem now being shifted for temporary silence in calls > where one party could not hear the other. This lasts for even 2 to 2.5 > seconds. I got 2 * server where one is connected to PSTN and the other to > internal PBX. When calls are from extension to the outside, it flows > like................ extension->pbx---ISDN > PRIE1---->server2----IAX2--->server1-----ISDN PRIE1--->PSTN. > Both servers are in the same LAN. > I've got tos=reliability > Does jitter has to do anything here. I've got my jitter set to default. > I'll send you a debug span in time.
Jitter buffer is probably the culprit now. Turn jitter off and you should have no problems. This is what occurred to us on our 2 * servers with a T1 data link in between. > ----- Original Message ----- > From: "Martin Pycko" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Wednesday, September 17, 2003 9:20 PM > Subject: Re: [Asterisk-Users] calls terminating abnormally > > > > Can you send a "pri debug span <span_no>" trace ? Or do you have an analog > > T1/E1 ? > > > > regards > > Martin > > > > On Wed, 17 Sep 2003, denzel-infotechs wrote: > > > > > hi! > > > I've got a asterisk system running with around 50 per calls per > minute. I've connected * to internal pabx and outside telecom using E1 > (ISDN pris). Sometimes calls disconect abnormally. Is this something we have > to live with or is it a bug in CVS code ? > > > > > > denzel. > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users